Jared Smith wrote: > On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org > wrote: > >> Randomly I have dropped calls during communication. No absolutetimeout or >> other >> calling limitation options. >> >> Any ideas on how to solve this problem? >> > > The first place I'd look would be the Asterisk CLI. Make sure you turn > up the CLI verbosity first by typing "core set verbose 5" before the > call. If that doesn't offer any clues, I'd next look at the SIP > signaling. You can see that by typing "sip set debug" at the Asterisk > CLI, or by using a network tool such as Wireshark, tcpdump, or ngrep. > > --- > Jared Smith > Community Relations Manager > Digium, Inc. > > I would also ask that all user's keep a log or send an email to you with their extension, if the call was internal or external, time, and how long into the call that it dropped. Collecting this data might help you figure out a trend. I would open an SSH session with txt logging and ask everyone to submit a dropped call report and see if you can link up some common events or errors. You may find it is only happening on external calls which may look like a normal hangup and could indicate a problem with your PSTN connectivity.
Thanks, Steve Totaro _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
